On Mar 28, 2009, at 4:05 AM, Anthony Minessale wrote:
if you prefix the sofia dial string with sip: you should be able to
pass anything you want.
sofia/internal/sip:9085551212;npdi=yes;[email protected]:5060
Is that similar as this?
<action application="bridge" data="sofia/sip/[email protected]
"/>
got it from wiki:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridgecall#From_the_Dialplan
2009/3/27 James H Thompson <[email protected]>
Calls would be sent to the IP address after the '@' in the URI.
Section 19.1.1 of RFC 3261 seems to say that TEL URIs can be used as
the user part of a SIP URI.
My example Invite URI is the way we are receiving traffic from some
of the major telecom carriers.
We would like be able to generate calls using the same formats.
----- Original Message -----
From: Michael Jerris
To: [email protected]
Sent: Friday, March 27, 2009 5:41 AM
Subject: Re: [Freeswitch-users] generating RFC 3966 and RFC 4694 calls
You seem to be confusing your standards, those 2 specs are about
tel: uri's not sip: uris. Sending a tel uri I am not sure we can
do, where would we send it to?
Mike
On Mar 27, 2009, at 6:11 AM, James H Thompson wrote:
I need to generate calls with Invite URIs in this format:
INVITE sip:9085551212;npdi=yes;[email protected]:5060
SIP/2.0
Is there an easy way to do this?
Thanks.
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--
Anthony Minessale II
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